[asterisk-bugs] [JIRA] (ASTERISK-23425) No sound when i make call from chrome via webrtc (sipml5) to asterisk extension. Asterisk return answer without ice-ufrag and ice-pwd.

[Kirill Ushakov (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-23425?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=217045#comment-217045 ] 

Kirill Ushakov commented on ASTERISK-23425:
-------------------------------------------

Hello MAX.
Thank you for your answer. 

Webrtc implemented in browser and if there some changes, asterisk or media gateway must implement them too.
Media gateway (if you mean webrtc2sip) developers update his own software faster than asterisk developers?  
I'm little don't understand where is profit here.

I think much better if asterisk developers watch your link and prepare asterisk for this changes :)

Best regards, Kirill

> No sound when i make call from chrome via webrtc (sipml5) to asterisk extension. Asterisk return answer without ice-ufrag and ice-pwd.
> --------------------------------------------------------------------------------------------------------------------------------------
>
>                 Key: ASTERISK-23425
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-23425
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/SRTP, Channels/chan_sip/WebSocket, Resources/res_http_websocket, Resources/res_srtp
>    Affects Versions: 11.4.0, 11.7.0, 11.8.0, 11.8.1
>         Environment: ********** OS ***********
> Distributor ID: Ubuntu
> Description:    Ubuntu 12.04.3 LTS
> Release:        12.04
> Codename:       precise
> **********kernel*************
> Linux version 3.8.0-29-generic (buildd at panlong) (gcc version 4.6.3 (Ubuntu/Linaro 4.6.3-1ubuntu5) ) #42~precise1-Ubuntu SMP Wed Aug 14 16:19:23 UTC 2013  x86_64 GNU/Linux
> **********Chrome************
> Chrome 33.0.1750.146 m
> **********Asterisk************
> Astersik version 11.8.0 with SRTP ( ./configure CFLAGS=-fPIC --prefix=/usr ) configured.
> ***Asterisk config users.conf*** 
> [4343]
> canreinvite = no
> type = peer
> host = dynamic
> context = mycontext
> hassip = yes
> hasiax = no
> nat = force_rport,comedia
> qualify = no
> encryption = yes
> avpf = yes
> ;savpf = yes
> language = ru
> videosupport = no
> directmedia = no
> disallow = all
> allow = alaw
> secret = mysecret
> transport = ws,udp
> icesupport = yes
> ***called extention in extensions.ael***
> 7 => {
>      Answer();
>      Playback(hello-world);
>      MusicOnHold(default,300);
>      Hangup();
>      }
>            Reporter: Kirill Ushakov
>            Assignee: Kirill Ushakov
>              Labels: asterisk, sdp
>         Attachments: asterisk_output.txt, chrome_output.txt, extensions.ael.txt, jira-jssip.txt, jira-sipdebug.txt, jira-sipml5.txt, os_kernel_asterisk_chrome.conf, rtp.conf.txt, sip.conf.txt, sipconf.txt
>
>
> No sound when i make call from chrome via webrtc (sipml5) to asterisk extension. Asterisk return answer without ice-ufrag and ice-pwd.
> In attachments i change my "External IP" and "Domain" to "195.195.195.195"  and "my.domain.com"
> for reproduce the problem, your server must be behind the NAT.



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